The present invention relates to systems and methods for improving the quality of a call over network through the optimization of packet delivery over the internet and other networks. Such systems and methods enable more efficient communications over networks than is currently available due to better communication fidelity. This results in faster and more productive calls.
Currently, a number of platforms are available for call over network (CON) communications. These typically include audio and/or video calling capabilities that rely upon the internet (or other suitable network) in order to enable communications between two or more participants. Examples of current CON systems include Vonage, Skype, WebEx, and Facetime, for example. Each of these systems have some differentiating features, but predominantly operate via scheduling or otherwise initiating a call, and then transmitting and receiving audio and/or video material via the internet (and often other networks as well). In some of the more sophisticated CON systems, such as WebEx, participants additionally have the capability to share their computer desktop, and further, pass this functionality to other participants.
While these CON systems have done a lot to enhance causal communications, the larger impact of CON systems is arguably on in relation to how businesses operate. Traditionally, in-face meetings were required to fully engage other business partners. The conference call was available, become costly when used for international calls since they operate over traditional telecommunications infrastructures. These newer CON systems have further increased how easily remote participants can communicate effectively; however, there are still a number of problems that tend to plague existing CON systems.
For example, proper connectivity of all users in a CON system is routinely an issue. Often one participant can have trouble joining or hearing without the other participant's knowledge. Connectivity issues are often sources of inefficiencies for call over network communications. Indeed, one of the largest problems facing CON systems is the fact that data is typically transmitted via the internet, which is a “best effort network”. Best effort network means that the data is transmitted with the best timing and quality reasonably possible. However, the data is transmitted over often torturous pathways, in sub-optimal conditions. As such, often timing and/or quality of the transmitted data are negatively impacted. Given that audio and video communications are done in real-time, these types of data transfers are very susceptible to transmission delays and/or packet loss.
Traditional call over network systems handle this reduction in call quality and/or timing by reducing high data demanding communications. For example, in Skype, the video portion of the call may have a reduced quality, or may be halted altogether. Additionally, these existing systems simply drop the call if the timing and/or quality gets below a threshold. The thinking behind these dropped calls is that the inconvenience of not being able to communicate is less burdensome than a bad connection.
While there is some merit to this business model, there are some circumstances where communication is required, even in sub-optimal network conditions. This can be especially true where a number of participants are engaging in a conference call. Here schedules are often difficult to coordinate, and as such the need to communicate via that medium, and at that time, are magnified.
Typically, when the participants are located relatively near one another, in a location with a decent internet backbone, these measures are sufficient to ensure basic levels of communication quality. However, when participants are more remotely located (for example on different continents), these measures may simply be insufficient in order to produce decent communication.
Indeed, for a decent voice over internet protocol (VoIP) call, total latency should be less than 400 ms, and have a packet loss of less than 2%. In contrast, the typical jitter for data between China and the United States can vary from a few hundred milliseconds to a few seconds. This is in addition to a 100-160 ms latency. Packet loss between China and the United States often varies from 0 to 100%. Clearly, the internet infrastructure is sub-optimal for communications from China to the US. Other global locations vary in their suitability for communication as well.
All of these drawbacks to existing CON systems require that callers repeat information more often, and reduce efficiency for all members. Moreover, in the extreme situation of a badly compromised network connection, existing CON systems are rendered inoperable. Impatient participants that have a good connection may quickly lose interest in the conversation as the pace seems unbearably slow, or as the calls are dropped. Other participants may leave the call missing much of what has been communicated. Often these participants don't want to ask for clarification too often due to respect for the numerous other participants' time. In the case of dropped calls, important communication may simply never happen.
It is therefore apparent that an urgent need exists for systems and methods for improving communication packet delivery over a network in order to improve the quality of a call over network. Such systems and methods provide optimization of the internet backbone, optimization of last mile delivery, and real time monitoring and mitigation technologies in order to achieve the best possible transmission outcomes given the infrastructure limitations present.